Older/MediaServer/Rtcp/RtcpContext.h
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add ZLMediaKit code for learning.
2024-09-28 23:55:00 +08:00

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/*
* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
*
* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
*
* Use of this source code is governed by MIT-like license that can be found in the
* LICENSE file in the root of the source tree. All contributing project authors
* may be found in the AUTHORS file in the root of the source tree.
*/
#ifndef ZLMEDIAKIT_RTCPCONTEXT_H
#define ZLMEDIAKIT_RTCPCONTEXT_H
#include "Rtcp.h"
#include <stddef.h>
#include <stdint.h>
namespace mediakit {
class RtcpContext {
public:
using Ptr = std::shared_ptr<RtcpContext>;
virtual ~RtcpContext() = default;
/**
* 输出或输入rtp时调用
* @param seq rtp的seq
* @param stamp rtp的时间戳单位采样数(非毫秒)
* @param ntp_stamp_ms ntp时间戳
* @param rtp rtp时间戳采样率视频一般为90000音频一般为采样率
* @param bytes rtp数据长度
* Called when outputting or inputting rtp
* @param seq rtp's seq
* @param stamp rtp's timestamp, unit is sample number (not millisecond)
* @param ntp_stamp_ms ntp timestamp
* @param rtp rtp timestamp sampling rate, video is generally 90000, audio is generally sampling rate
* @param bytes rtp data length
* [AUTO-TRANSLATED:745772b5]
*/
virtual void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes);
/**
* 输入sr rtcp包
* @param rtcp 输入一个rtcp
* Input sr rtcp packet
* @param rtcp input an rtcp
* [AUTO-TRANSLATED:46f309ec]
*/
virtual void onRtcp(RtcpHeader *rtcp) = 0;
/**
* 计算总丢包数
* Calculate the total number of lost packets
* [AUTO-TRANSLATED:084f3832]
*/
virtual size_t getLost();
/**
* 返回理应收到的rtp数
* Return the number of rtp that should be received
* [AUTO-TRANSLATED:ede367a0]
*/
virtual size_t getExpectedPackets() const;
/**
* 创建SR rtcp包
* @param rtcp_ssrc rtcp的ssrc
* @return rtcp包
* Create SR rtcp packet
* @param rtcp_ssrc rtcp's ssrc
* @return rtcp packet
* [AUTO-TRANSLATED:a9ec36d0]
*/
virtual toolkit::Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc);
/**
* @brief 创建xr的dlrr包用于接收者估算rtt
*
* @return toolkit::Buffer::Ptr
* @brief Create xr's dlrr packet, used by receiver to estimate rtt
*
* @return toolkit::Buffer::Ptr
* [AUTO-TRANSLATED:a5094e1d]
*/
virtual toolkit::Buffer::Ptr createRtcpXRDLRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
/**
* 创建RR rtcp包
* @param rtcp_ssrc rtcp的ssrc
* @param rtp_ssrc rtp的ssrc
* @return rtcp包
* Create RR rtcp packet
* @param rtcp_ssrc rtcp's ssrc
* @param rtp_ssrc rtp's ssrc
* @return rtcp packet
* [AUTO-TRANSLATED:81ebbf81]
*/
virtual toolkit::Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
/**
* 上次结果与本次结果间应收包数
* Number of packets that should be received between the last result and the current result
* [AUTO-TRANSLATED:3b2846ab]
*/
virtual size_t getExpectedPacketsInterval();
/**
* 上次结果与本次结果间丢包个数
* Number of lost packets between the last result and the current result
* [AUTO-TRANSLATED:fe5ac890]
*/
virtual size_t getLostInterval();
protected:
// 收到或发送的rtp的字节数 [AUTO-TRANSLATED:a38d88a9]
// Number of bytes of rtp received or sent
size_t _bytes = 0;
// 收到或发送的rtp的个数 [AUTO-TRANSLATED:b28c3c90]
// Number of rtp received or sent
size_t _packets = 0;
// 上次的rtp时间戳,毫秒 [AUTO-TRANSLATED:99eecec6]
// Last rtp timestamp, milliseconds
uint32_t _last_rtp_stamp = 0;
uint64_t _last_ntp_stamp_ms = 0;
};
class RtcpContextForSend : public RtcpContext {
public:
toolkit::Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc) override;
void onRtcp(RtcpHeader *rtcp) override;
toolkit::Buffer::Ptr createRtcpXRDLRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override;
/**
* 获取rtt
* @param ssrc rtp ssrc
* @return rtt,单位毫秒
* Get rtt
* @param ssrc rtp ssrc
* @return rtt, unit is millisecond
* [AUTO-TRANSLATED:f0885551]
*/
uint32_t getRtt(uint32_t ssrc) const;
private:
std::map<uint32_t /*ssrc*/, uint32_t /*rtt*/> _rtt;
std::map<uint32_t /*last_sr_lsr*/, uint64_t /*ntp stamp*/> _sender_report_ntp;
std::map<uint32_t /*ssrc*/, uint64_t /*xr rrtr sys stamp*/> _xr_rrtr_recv_sys_stamp;
std::map<uint32_t /*ssrc*/, uint32_t /*last rr */> _xr_xrrtr_recv_last_rr;
};
class RtcpContextForRecv : public RtcpContext {
public:
void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) override;
toolkit::Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override;
size_t getExpectedPackets() const override;
size_t getExpectedPacketsInterval() override;
size_t getLost() override;
size_t getLostInterval() override;
void onRtcp(RtcpHeader *rtcp) override;
private:
// 时间戳抖动值 [AUTO-TRANSLATED:8100680c]
// Timestamp jitter value
double _jitter = 0;
// 第一个seq的值 [AUTO-TRANSLATED:d893719d]
// The value of the first seq
uint16_t _seq_base = 0;
// rtp最大seq [AUTO-TRANSLATED:5cc9f775]
// Maximum rtp seq
uint16_t _seq_max = 0;
// rtp回环次数 [AUTO-TRANSLATED:9fe9c340]
// Rtp loopback times
uint16_t _seq_cycles = 0;
// 上次回环发生时记录的rtp包数 [AUTO-TRANSLATED:c32cb555]
// Number of rtp packets recorded when the last loopback occurred
size_t _last_cycle_packets = 0;
// 上次的seq [AUTO-TRANSLATED:07364b7d]
// Last seq
uint16_t _last_rtp_seq = 0;
// 上次的rtp的系统时间戳(毫秒)用于统计抖动 [AUTO-TRANSLATED:b1e8c89b]
// Last rtp system timestamp (milliseconds) used for jitter statistics
uint64_t _last_rtp_sys_stamp = 0;
// 上次统计的丢包总数 [AUTO-TRANSLATED:242e75ed]
// Last total number of lost packets counted
size_t _last_lost = 0;
// 上次统计应收rtp包总数 [AUTO-TRANSLATED:eb2d5f4d]
// Last total number of rtp packets that should be received counted
size_t _last_expected = 0;
// 上次收到sr包时计算出的Last SR timestamp [AUTO-TRANSLATED:fdec069e]
// Last SR timestamp calculated when the last SR packet was received
uint32_t _last_sr_lsr = 0;
// 上次收到sr时的系统时间戳,单位毫秒 [AUTO-TRANSLATED:044fa0d5]
// System timestamp when the last SR was received, unit is millisecond
uint64_t _last_sr_ntp_sys = 0;
};
} // namespace mediakit
#endif // ZLMEDIAKIT_RTCPCONTEXT_H